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Hello, I have one question regarding *. Default configuration for asterisk is to keep configuration(s) in ordinary text based config files. My question is simple: is it possible to keep those config info (at least, to start from - sip.conf, extensions.conf and voicemail.conf) on a database, which asterisk access via ODBC. If it is possible, I'd appreciate if someone points me where I can

Is there a way to display registered SIP useragents and sort them from CLI? I.N.

Hmmm... Folks, I beg you pardon, if I'm telling something which was said before, but actually I have not found this anywhere, neither on Voip-info.org or in several Asterisk's docs. So, here is the statement: If SIP extensions are in DIFFERENT CONTEXTS, then RTP traffic between them will ALWAYS go via Asterisk. I.e. Asterisk WILL NOT issue Re-INVITE even if: 1. Both UAs have

Hello, Question #1: I have * with g729 installed, and two phones - Cisco 7960 and Cisco 7905. If g729 is the only available codec for 7905's configuration, then call from 7960 to 7905 goes without any problem and both phones use g729. But if I call from 7905 to 7960 the following is displayed on * console: WARNING[5220]: rtp.c:1545 ast_rtp_bridge: codec0 = 256 is not codec1 = 4, cannot

I've got * registered with 50 SIP extensions. There are two another SIP proxies. I'd like to configure following: 1. Call from outside comes on *. * looks up for an extension 2. If no registered extension is on *, then request is forwarded to SIP proxy 1. 3. If client in not found on SIP Proxy 1, then * forwards request to SIP Proxy2 4. If client is not found SIP Proxy 2 congestion

Hello folks, I've did some tests with different phones and Asterisk last two days and here are some results, which I want to share with audience. Cisco's 79xx and Sipura's phones/adapters on INVITE always reply with their preferred codec. So, for example, if Cisco's/Sipura's phone has preferred_codec g729a(18) and it receives INVITE from UA which has preferred codec

Hello everybody, Do I understand correctly that Asterisk does not support sending INFO request? Here is the goal I want to accomplish and I'd be happy to hear how can it be done with asterisk. Asterisk needs to dial out and after successful call establishment it needs to send in-dialog INFO request to the callee and wait after that for another INFO message coming from callee. So call flow